1. Field of the Invention
The present invention relates to a mobile communication system, and more particularly, to voice communication in the mobile communication system.
2. Description of the Related Art
FIG. 5 shows a configuration of a conventional general communication system. Referring to FIG. 5, a mobile network 51 is interconnected with a fixed network 52 via a gateway (not shown).
The mobile network 51 is a mobile communication system constituted by a plurality of types of apparatuses including base stations and connected with mobile terminals 53 via radio links. The mobile communication system may have a variety of configurations. A signal is sent and received with a radio wave between the base station and the mobile terminal 53. The fixed network 52 is, for example, PSTN (public switched telephone network) or ISDN (integrated services digital network), and is connected with fixed telephones 54.
In a mobile communication system, frequency usage efficiency is improved wherever possible in order to achieve communications at mobile terminals of many users with a finite radio frequency resource. In a digital mobile communication system in which a voice is transmitted with a digital signal, a required frequency band can be narrowed if the transmission rate of each voice communication is reduced. Therefore, in the digital mobile communication system, it is desirable that each voice communication is coded in a mode of low transmission rate for improving the frequency usage efficiency. On the other hand, since voice quality is compromised if the transmission rate is reduced, the transmission rate may be increased for improving the voice quality when speech channels are not congested. Thus, if the transmission rate of each voice communication can be selected according to the degree of congestion, the frequency usage efficiency can be improved while the speech quality can be ensured at the same time.
In addition, in a mobile communication system, the quality of communication is generally degraded when the speed at which a mobile terminal moves is increased. Thus, it is desirable that the transmission rate of voice communication can adaptively be changed after the voice communication is started.
For this purpose, some conventional mobile communication systems employ an AMR (Adaptive Multi-Rate Codec) mode. In the AMR mode, the mobile communication system and the mobile terminal can use voice coding modes of a plurality of transmission rates, the network side of the mobile communication system and the mobile terminal determine the transmission rate by negotiation at the time when voice communication is started, and the transmission rate is changed to an appropriate rate according to the communication quality during voice communication.
A general procedure in which the mobile communication system employing the AMR mode changes the transmission rate during voice communication will be described.
First, an apparatus monitoring the quality of radio communication on the network side of the mobile communication system (hereinafter referred to as monitoring apparatus) determines whether or not the transmission rate should be changed according to the quality. If determining that the transmission rate should be changed, the monitoring apparatus instructs the mobile terminal to change the uplink transmission rate. The mobile terminal changes to an indicated rate the transmission rate at which signals are sent in the uplink.
The monitoring apparatus then instructs an apparatus having a voice codec (CODEC: Coder/Decoder) on the network side of the mobile communication system (hereinafter referred to as switching apparatus) to change the downlink transmission rate. When the switching apparatus changes the downlink transmission rate, the procedure of changing the transmission rate is completed.
For one example of control of the transmission rate, the transmission rate is increased when the quality of radio communication is degraded, and the transmission rate is decreased when speech channels are congested.
The IMT-2000 (International Mobile Telecommunications-2000) system that was made available in 2001 in Japan and the GSM (Global Systems for Mobile communications) system operated mainly in Europe employ the AMR mode.
FIG. 6 is a sequence diagram showing a procedure for changing the transmission rate in the IMT-2000 system. In the IMT-2000 system, a CN (Core Network) constituted by a plurality of MSCs (Mobile services Switching Centers), a RNC (Radio Network Controller) connected to MSC and UE (User Equipment) send and receive control signals, whereby the transmission rate is controlled. The MSC is an exchange constituting the mobile communication system. Since the MSC has a voice codec, it is equivalent to the above described switching apparatus, and switches between the voice coding mode of the mobile communication system and the PCM (pulse code modulation) that is used in PSTN or the like. The RNC is connected to a plurality of base stations, and controls the base stations and radio links. The RNC is equivalent to the above described monitoring apparatus. The UE is a terminal of a cellular phone or the like, and has a voice codec.
For the procedure of changing the transmission rate, the RNC sends to the UE a control signal providing an instruction to change the uplink transmission rate [RATE CONTROL FOR UPLINK] when the RNC determines that the transmission rate should be changed based on the quality of radio communication.
Upon reception of the instruction to change the uplink transmission rate, the UE starts sending a voice signal as user data at the indicated transmission rate.
Then, the RNC sends to a predetermined MSC of the CN a control signal providing an instruction to change the downlink transmission rate [RATE CONTROL FOR DOWNLINK].
Upon reception of the instruction to change the downlink transmission rate, the MSC starts sending a voice signal as user data at the indicated transmission rate.
FIG. 7 is a sequence diagram showing a procedure for changing the transmission rate in the GSM system. In the GSM system, a TRAU (Transcoder Rate Adaptation Unit), a BTS (Base Transceiver Station) and UE (User Equipment) send and receive control signals, where by the transmission rate is controlled. The TRAU is provided between a BSC (Base Station Controller) controlling the BTS and the BTS, and switches between the voice coding mode of the mobile communication system and the PCM. The TRAU is equivalent to the above described switching apparatus. The BSC is connected to a CN (Cellular Network) or PSTN. The BTS is a base station for carrying out radio communication with the UE, and is equivalent to the above described monitoring apparatus. The UE is a terminal of a cellular phone or the like, and has a voice codec.
For the procedure of changing the transmission rate, the BTS sends to the UE a control signal providing an instruction to change the uplink transmission rate [RATE CONTROL FOR UPLINK] when the BTS determines that the transmission rate should be changed based on the quality of radio communication.
When instructed to change the uplink transmission rate, the UE starts sending a voice signal as user data at the indicated transmission rate.
Then, the BTS sends to the BSC a control signal providing an instruction to change the downlink transmission rate [RATE CONTROL FOR DOWNLINK].
When instructed to change the downlink transmission rate, the BSC starts sending a voice signal as user data at the indicated transmission rate.
In the GSM system and IMT-2000 system, the transmission rate is adaptively controlled according to the quality of radio communication and the degree of congestion of the speech channels, thus making it possible to perform satisfactory communication.
The conventional mobile communication system such as the GSM system and IMT-2000 system includes an apparatus having a voice codec, and a voice signal passing through a gateway is of PCM when a mobile terminal makes voice communication with the terminal of a fixed network such as PSTN and ISDN. The protocol for controlling the transmission rate of the mobile communication system is terminated in the mobile communication system.
In recent years, on the other hand, the IP telephone for making voice communication by using a terminal connected to the IP network has been spread more and more. The IP network accommodates the terminal of the personal computer or the like by the Internet protocol, and is interconnected with the mobile network and the fixed network.
In the IP telephone, a voice coding mode for use in the fixed telephone such as the PCM or ADPCM (adaptive differential PCM) may be used, but it is desirable that a voice coding mode of low transmission rate is used as in the case of mobile communication from a viewpoint of accommodation efficiency in speech channel.
If the IP telephone and the mobile communication system individually select (band-compressed) voice coding modes of low transmission rate to achieve voice communication between the mobile terminal and the IP telephone terminal, band compression and extension occurs twice for the voice signal in the mobile communication system and the IP network, resulting in degradation of voice quality. Thus, if the mobile terminal having a voice codec and an apparatus having a voice codec in the IP network directly send and receive a coded voice signal, the band compression and extension occurs only once, and therefore the degradation of voice quality is alleviated.
In this case, however, the apparatus controlling the voice coding mode of the mobile communication system does not have a protocol for exchanging control signals for changing the voice coding mode between itself and the apparatus in the IP network, and therefore the voice coding mode cannot be changed during voice communication. In order that the mobile communication system and the IP network change the voice coding mode in connection with each other, a control protocol for controlling the voice coding mode may be defined between the apparatus controlling the voice coding mode of the mobile communication system and the apparatus in the IP network. However, it can be considered that there exist a variety of mobile communication systems having different configurations such as the GSM system and the IMT-2000 system. Thus, the apparatus in the IP network should operate in a different way for each mobile communication system because its partner to which the control signal is sent and from which the control signal is received for controlling the voice coding mode is different for each mobile communication system.
If a mobile communication system having further diverse configurations appears hereafter, additional processing should be applied to the apparatus in the IP network, or an apparatus should be added to the IP network on each such occasion, thus raising development and equipment costs.